/*
 ____  _____ _        _    
| __ )| ____| |      / \   
|  _ \|  _| | |     / _ \  
| |_) | |___| |___ / ___ \ 
|____/|_____|_____/_/   \_\

The platform for ultra-low latency audio and sensor processing

http://bela.io

A project of the Augmented Instruments Laboratory within the
Centre for Digital Music at Queen Mary University of London.
http://www.eecs.qmul.ac.uk/~andrewm

(c) 2016 Augmented Instruments Laboratory: Andrew McPherson,
  Astrid Bin, Liam Donovan, Christian Heinrichs, Robert Jack,
  Giulio Moro, Laurel Pardue, Victor Zappi. All rights reserved.

The Bela software is distributed under the GNU Lesser General Public License
(LGPL 3.0), available here: https://www.gnu.org/licenses/lgpl-3.0.txt
*/


#include <Bela.h>
#include <rtdk.h>
#include <cmath>

float gPhase;
float gInverseSampleRate;
int gAudioFramesPerAnalogFrame;

// These settings are carried over from main.cpp
// Setting global variables is an alternative approach
// to passing a structure to userData in setup()

extern int gSensorInputFrequency;
extern int gSensorInputAmplitude;

bool setup(BelaContext *context, void *userData)
{
	if(context->analogFrames == 0 || context->analogFrames > context->audioFrames) {
		rt_printf("Error: this example needs analog enabled, with 4 or 8 channels\n");
		return false;
	}

	gAudioFramesPerAnalogFrame = context->audioFrames / context->analogFrames;
	gInverseSampleRate = 1.0 / context->audioSampleRate;
	gPhase = 0.0;

	return true;
}

void render(BelaContext *context, void *userData)
{
	float frequency = 440.0;
	float amplitude = 0.8;

	// There are twice as many audio frames as matrix frames since audio sample rate
	// is twice as high

	for(unsigned int n = 0; n < context->audioFrames; n++) {
		if(!(n % gAudioFramesPerAnalogFrame)) {
			// Even audio samples: update frequency and amplitude from the matrix
			frequency = map(analogRead(context, n/gAudioFramesPerAnalogFrame, gSensorInputFrequency), 0, 1, 100, 1000);
			amplitude = analogRead(context, n/gAudioFramesPerAnalogFrame, gSensorInputAmplitude);
		}

		float out = amplitude * sinf(gPhase);

		for(unsigned int channel = 0; channel < context->audioChannels; channel++)
			context->audioOut[n * context->audioChannels + channel] = out;

		gPhase += 2.0 * M_PI * frequency * gInverseSampleRate;
		if(gPhase > 2.0 * M_PI)
			gPhase -= 2.0 * M_PI;
	}
}

void cleanup(BelaContext *context, void *userData)
{

}

/* ------------ Project Explantation ------------ */

/**
\example 03-analog-input

Connecting potentiometers
-------------------------

This sketch produces a sine tone, the frequency and amplitude of which are 
affected by data received on the analog pins. Before looping through each audio 
frame, we declare a value for the frequency and amplitude of our sine tone 
(line 55); we adjust these values by taking in data from analog sensors 
(for example potentiometers) with `analogRead()`.

- connect a 10K pot to 3.3V and GND on its 1st and 3rd pins.
- connect the 2nd middle pin of the pot to analogIn 0.
- connect another 10K pot in the same way but with the middle pin connected to analogIn 1.

The important thing to notice is that audio is sampled twice as often as analog 
data. The audio sampling rate is 44.1kHz (44100 frames per second) and the 
analog sampling rate is 22.05kHz (22050 frames per second). On line 62 you might 
notice that we are processing the analog data and updating frequency and 
amplitude only on every second audio sample, since the analog sampling rate is 
half that of the audio.

Note that the pin numbers are stored in the variables `gAnalogInputFrequency` and 
`gAnalogInputAmplitude`. These are declared in the main.cpp file; if you look in 
that file you will see that they have the values of 0 and 1. Bear in mind that 
these are analog input pins which is a specific header!
*/
